This one had to go a little more basic to What is SIP? SIP Trunking is how carriers power dial-tone to an IP-PBX. Beyond the definition of the term, there is the concept that this specification for a voice packet to provide dial-tone is the foundation of the Next-Gen Communication platform. Start with the SIP trunk as the basic circuit needed for communications. Then other applications that utilize SIP sessions can be integrated into the platform. (In many cases, the platform is the PBX or the softswitch.)

One point that I made: All voice will soon be Voice over IP. Cellular is. Long Distance is. Dynamic T1 is. Cableco digital voice is. AT&T is turning off the PSTN with the approval of the FCC in less than 10 years. SIP has become the de facto protocol for Voice over IP - beating H.323, MGCP and others. Agents (and customers) have to get on-board with this fact. SIP is here to stay.

Yes there are issues. Most notably Fax-over-IP (which many companies are working on, T.38 not withstanding); HD Voice; alarms, elevators, credit card processing machines; and inter-operability between the SIP provider and the gear attached to the Trunk. Since SIP is a specification of about 30 RFC's, it is interpreted differently by manufacturers and providers. Hence, inter-operability is extremely important. It is not like a PRI, which is a standard, with just 2 configurations available in any class 4 or 5 switch or any PBX system. That makes inter-op easy. Today. Maybe not so much in 1988.

Our panel then went a little crazy talking about Unified Communication and all the possible UC components that could be mounted on that SIP trunk - like SMS/text, IM/chat, Video, Presence, conferencing, ACD, IVR, etc. Unfortunately, it's not that easy and most of the integration is with the gear (PBX, IAD, softswitch, SBC).

During Q&A, some asked how to explain SIP to a customer. I said, Don't. Would you explain how the engine works in a car? Focus on the benefits, the reliability, the way it will help the business and you won't have to worry about saying VoIP or SIP.

The panel had to explain that SIP trunks mean different things to different carriers. In some cases, a SIP trunk is just a call path. In other cases, a SIP Trunk is a circuit containing many call paths. It's confusing for the agent and the customer.

There are two ways to deliver SIP Trunking: over an IP circuit and over the Internet. Big difference. SIP sessions over the Internet lack quality of service. Jitter and latency will affect the voice quality. SIP Trunks that ride private line, virtual circuit or MPLS circuit have not only the best call quality but enhanced security.

Many ITSP's started out as ISP's (or at least with an ISP infrastructure). At the NOC, the ITSP has aggregation circuits for DSL, MPLS, private line from ILEC's and CLEC's. Some even have aggregation with a cableco. This means that the ITSP - in its network footprint - is delivering VoIP service to you on a private network with QOS of some level. The customer is receiving Internet over the same circuit but the egress to The Web is AFTER the aggregation point and separate from the softswitch.

Major MSO's deliver their Enterprise VoIP product as a physically separate VLAN. That offers QOS and security as well.

At XO, Pete Davis and I have collaborated on a concept called The UC Sandbox. XO has all the components for UC available for sale. Hosted MS Exchange, Blackberry server, Hosting, Collocation, IM/chat, Hosted PBX, SIP Trunking, Contact center application, web & audio conferencing, video conferencing, data storage, Anywhere service, and more can be provisioned for the customer as a single individual service or as a bundle of components. The bonus for VAR's is that if they have a business selling email, security, what-have-you, they can mix-and-match components from XO or from a variety of providers to put the best solution together for the customer. That is the true value of VoIP, SIP and the future of Cloud.

This one had to go a little more basic to What is SIP? SIP Trunking is how carriers power dial-tone to an IP-PBX. Beyond the definition of the term, there is the concept that this specification for a voice packet to provide dial-tone is the foundation of the Next-Gen Communication platform. Start with the SIP trunk as the basic circuit needed for communications. Then other applications that utilize SIP sessions can be integrated into the platform. (In many cases, the platform is the PBX or the softswitch.)

\n

One point that I made: All voice will soon be Voice over IP. Cellular is. Long Distance is. Dynamic T1 is. Cableco digital voice is. AT&T is turning off the PSTN with the approval of the FCC in less than 10 years. SIP has become the de facto protocol for Voice over IP - beating H.323, MGCP and others. Agents (and customers) have to get on-board with this fact. SIP is here to stay.

\n

Yes there are issues. Most notably Fax-over-IP (which many companies are working on, T.38 not withstanding); HD Voice; alarms, elevators, credit card processing machines; and inter-operability between the SIP provider and the gear attached to the Trunk. Since SIP is a specification of about 30 RFC's, it is interpreted differently by manufacturers and providers. Hence, inter-operability is extremely important. It is not like a PRI, which is a standard, with just 2 configurations available in any class 4 or 5 switch or any PBX system. That makes inter-op easy. Today. Maybe not so much in 1988.

\n

Our panel then went a little crazy talking about Unified Communication and all the possible UC components that could be mounted on that SIP trunk - like SMS/text, IM/chat, Video, Presence, conferencing, ACD, IVR, etc. Unfortunately, it's not that easy and most of the integration is with the gear (PBX, IAD, softswitch, SBC).

\n

During Q&A, some asked how to explain SIP to a customer. I said, Don't. Would you explain how the engine works in a car? Focus on the benefits, the reliability, the way it will help the business and you won't have to worry about saying VoIP or SIP.

\n

The panel had to explain that SIP trunks mean different things to different carriers. In some cases, a SIP trunk is just a call path. In other cases, a SIP Trunk is a circuit containing many call paths. It's confusing for the agent and the customer.

\n

There are two ways to deliver SIP Trunking: over an IP circuit and over the Internet. Big difference. SIP sessions over the Internet lack quality of service. Jitter and latency will affect the voice quality. SIP Trunks that ride private line, virtual circuit or MPLS circuit have not only the best call quality but enhanced security.

\n

Many ITSP's started out as ISP's (or at least with an ISP infrastructure). At the NOC, the ITSP has aggregation circuits for DSL, MPLS, private line from ILEC's and CLEC's. Some even have aggregation with a cableco. This means that the ITSP - in its network footprint - is delivering VoIP service to you on a private network with QOS of some level. The customer is receiving Internet over the same circuit but the egress to The Web is AFTER the aggregation point and separate from the softswitch.

\n

Major MSO's deliver their Enterprise VoIP product as a physically separate VLAN. That offers QOS and security as well.

\n

At XO, Pete Davis and I have collaborated on a concept called The UC Sandbox. XO has all the components for UC available for sale. Hosted MS Exchange, Blackberry server, Hosting, Collocation, IM/chat, Hosted PBX, SIP Trunking, Contact center application, web & audio conferencing, video conferencing, data storage, Anywhere service, and more can be provisioned for the customer as a single individual service or as a bundle of components. The bonus for VAR's is that if they have a business selling email, security, what-have-you, they can mix-and-match components from XO or from a variety of providers to put the best solution together for the customer. That is the true value of VoIP, SIP and the future of Cloud.